In an internet protocol (IP) interconnection environment, such as next generation network (NGN), data is exchanged between different personal computers (PCs) or terminals in business networks.
In data communication, a terminal or a PC is typically provided with software for opening files or data in a transmission mode and for supporting them. Accordingly, no problems generally occur with respect to simple data transmission between business networks (e.g., text, numbers, graphics, etc.).
However, in a communication operation using an IP interconnection, media data such as audio or video data may be included in addition to simple data. Currently there is a problem with IP interconnections that include media data because there is no standardized codec among providers. In other words, each different business network may use a preferred codec, which may be different from a preferred network used by another business network. For example, as shown in FIG. 2, a business network A may employ an ITU-T G.711 codec, a business network B may employ an enhanced variable rate codec (EVRC), and a business network C may employ an adaptive multi-rate (AMR) codec. In this case, a phone terminal in business network A and a phone terminal in business network B cannot be interconnected to each other without performing a proper codec conversion.
As a first method for solving this problem of codec conversion, it is possible that a phone terminal could be provided with a codec corresponding to each business network it might connect to. However, this provides both a physical problem and a technical problem regarding whether or not the large number of phone terminals that are already in use in the market can be provided with this additional function.
For this reason, a second method for solving the problem in codec conversion has been proposed in which codec conversion is performed at a network border (i.e., at a border between business networks). In this way, the same method used for codec conversion in a phone terminal can also be applied to codec conversion performed at a border between phone terminals.
FIG. 3 is a diagram illustrating a codec converting method in the phone terminals (providing a conversion from G.711 into EVRC). In this method, encoded audio data is extracted from a real-time transport protocol (RTP) packet having audio data encoded according to G.711 (T1); jitter is adjusted by using a jitter buffer (T2); and the encoded audio data is decoded (T3) to restore original audio data. The original audio data is temporarily stored in a reproduction buffer, where interpolation of packet loss is performed (T4); encoding is performed according to an encoding mode (EVRC) for the post-conversion side (T5), and the encoded audio data is again inserted to the RTP packet (T6).
However, this second method, in which the codec conversion is performed at a network border between business networks, also has problems. These problems include: (a) the fact that various kinds of codec conversion may not be supported, (b) that there is no countermeasure against a simultaneous process over multiple channels, and (c) that a conversion process delay is not considered.
The second method is a method contrived to easily install the code conversion function (See, FIG. 3) corresponding to communication between two terminals, into a small-scale gateway device. In such an implementation, the aforementioned problems (a) to (c) may occur.
There are many kinds of business networks relating to IP interconnection (and therefore to a large number of codecs). Accordingly, when many kinds of codec conversion are not supported in the existing gateway device, another gateway device needs to be installed for many kinds of codec conversion. Therefore, the system will become complicated and large in size.
In IP interconnection, since there are a large number of channels connected between business networks it is performable that a single codec conversion device can correspond to such a large number of channels.
Real-time processing in communication is also important even in IP interconnection, and a media transmission delay including a codec conversion processing time must be kept to a minimum. In certain communications requirements (e.g., provision set forth by the Japanese Ministry of Internal Affairs and Communications or the like), target end-to-end delays (i.e., for the connection between a terminal of the business network A and a terminal of the business network B in FIG. 2) of audio communication are set to be within 100 ms. Similarly, the target end-to-end delay for video communication is set to be within 200 ms. This target value is a delay that does not make a user feel unnatural during communication and conversation. It is determined from subjective measurement. When the delay substantially exceeds the regulated value, conversational communication may deteriorate significantly, and the user is placed under stress.
In the above description, the problems relating to audio data have generally been described. However, the same problems are equally applicable to video data.
For this reason, it is desirable to provide a codec converter, a gateway device, and a codec converting method, that are capable of coping with multiple kinds of codec conversion and the simultaneous processing of a plurality of channels, while having a low delay.